The present invention relates to digital communications, and more particularly to a method and apparatus for recovering carrier phase in an adaptive equalizer without the use of phase rotation or de-rotation.
Digital data, for example digitized video for use in broadcasting high definition television (HDTV) signals, can be transmitted over terrestrial very high frequency (VHF) or ultra high frequency (UHF) analog channels for communication to end users. Analog channels deliver corrupted and transformed versions of their input waveforms. Corruption of the waveform, usually statistical, may be additive and/or multiplicative, because of possible background thermal noise, impulse noise, and fades. Transformations performed by the channel are frequency translation, nonlinear or harmonic distortion, and time dispersion.
In order to communicate digital data via an analog channel, the data is modulated using, for example, a form of pulse amplitude modulation (PAM). Typically, quadrature amplitude modulation (QAM) is used to increase the amount of data that can be transmitted within an available channel bandwidth. QAM is a form of PAM in which a plurality of bits of information are transmitted together in a pattern referred to as a "constellation", which can contain, for example, sixteen or thirty-two points.
In pulse amplitude modulation, each signal is a pulse whose amplitude level is determined by a transmitted symbol. In 16-QAM, symbol amplitudes of -3, -1, 1 and 3 in each quadrature channel are typically used. In bandwidth efficient digital communication systems, the effect of each symbol transmitted over a time-dispersive channel extends beyond the time interval used to represent that symbol. The distortion caused by the resulting overlap of received symbols is called intersymbol interference (ISI). This distortion has been one of the major obstacles to reliable high speed data transmission over low background noise channels of limited bandwidth. A device known as an "equalizer" is used to deal with the ISI problem.
In order to reduce the intersymbol interference introduced by a communication channel, rather precise equalization is required. Furthermore, the channel characteristics are typically not known beforehand. Thus, it is common to design and use a compromise (or a statistical) equalizer that compensates for the average of the range of expected channel amplitude and delay characteristics. A least mean square (LMS) error adaptive filtering scheme has been in common use as an adaptive equalization algorithm for over 20 years. This algorithm is described in B. Widrow and M. E. Hoff, Jr., "Adaptive Switching Circuits" in IRE Wescon Conv. Rec., Part 4, pp. 96-104, Aug. 1960. The use of the LMS algorithm in an adaptive equalizer to reduce intersymbol interference is discussed in S. U. H. Qureshi, "Adaptive Equalization", Proc. IEEE, Vol. 73, No. 9, pp. 1349-1387, September 1987.
In an LMS equalizer, the equalizer filter coefficients are chosen to minimize the mean square error, i.e., the sum of squares of all the ISI terms plus the noise power at the output of the equalizer. Therefore, the LMS equalizer maximizes the signal-to-distortion ratio at its output within the constraints of the equalizer time span and the delay through the equalizer. Before regular data transmission begins, automatic synthesis of the LMS equalizer for unknown channels may be carried out during a training period. This generally involves the iterative solution of a set of simultaneous equations. During the training period, a known signal is transmitted and a synchronized version of the signal is generated in the receiver to acquire information about the channel characteristics. The training signal may consist of periodic isolated pulses or a continuous sequence with a broad, uniform spectrum such as a widely known maximum length shift register or pseudo-noise sequence.
An important aspect of equalizer performance is its convergence, which is generally measured by the amount of time in symbol periods required for the error variance in the equalizer to settle at a minimum level, which is ideally zero. In order to obtain the most efficient operation for a data receiver, the equalizer convergence time must be minimized.
After any initial training period, the coefficients of an adaptive equalizer may be continually adjusted in a decision directed manner. In this mode, the error signal is derived from the final receiver estimate (not necessarily correct) of the transmitted sequence. In normal operation, the receiver decisions are correct with high probability, so that the error estimates are correct often enough to allow the adaptive equalizer to maintain precise equalization. Moreover, a decision directed adaptive equalizer can track slow variations in the channel characteristics or linear perturbations in the receiver front end, such as slow jitter in the sampler phase.
Many transmission systems employ modulation schemes that are constructed with complex signal sets. In other words, the signals are viewed as vectors in the complex plane, with the real axis called the inphase (I) channel and the imaginary axis called the quadrature (Q) channel. Consequently, when these signals are subjected to channel distortion and receiver impairments, cross talk between the I and Q channels occurs, requiring a complex adaptive equalizer. In this case, the equalizer's coefficients will be complex valued. If, as noted above, the channel distortion is unknown by the receiver, then the coefficients must be adjusted after the system has been in operation to cancel the channel distortion. The term "adaptive" in a complex adaptive equalizer signifies the ongoing adjustment of the coefficients.
In many practical transmission systems, some method must be provided to derive a reference signal at the receiver's demodulator that is phase coherent with the received signal. Such coherent demodulators are used to demodulate signals containing information in their phase. For example, in binary phase shift keying (BPSK), modulation of a digital "one" is represented by a phase of zero degrees and modulation of a "zero" is represented by a phase of 180 degrees in the modulated signal. Data modulated using QAM techniques is demodulated on the basis of similar, although more complicated, phase relationships. Thus, demodulators for such data rely on a reference signal that must be synchronized in phase with the data carrier. This process is known as carrier phase recovery (CPR).
A phase locked loop (PLL) is a common and well known method used to recover the carrier in signal demodulators. When used in such applications, the PLL is sometimes referred to as a carrier recovery loop (CRL). When an adaptive equalizer is employed, it has been common practice to locate the CRL after the equalizer in the receiver. A free running oscillator is used to translate the input signal frequency to baseband, and a phase rotator is required to recover the carrier phase. In addition, a phase de-rotator is required in the adaptive equalizer to provide a correctly phased error signal for use in updating the filter coefficients. The requirement for a phase rotator and de-rotator complicates the receiver design, and adds expense to the receiver circuitry.
It would be advantageous to provide a method for recovering carrier phase in systems employing adaptive equalization without the need for phase rotation and de-rotation hardware. It would be further advantageous to provide an adaptive equalizer for a communications receiver that can initially adjust the equalizer coefficients in the absence of carrier phase recovery, thereby reducing the acquisition time of the system. Reduction of the system complexity by using self-recovering equalization algorithms that do not require a training sequence would be further advantageous. Such a system would be able to commence equalization without waiting for carrier recovery to occur.
The present invention provides a method and apparatus enjoying the aforementioned and other advantages.